sofia-sip-devel — List for discussions related to use and development of Sofia-SIP components

Re: [Sofia-sip-devel] Sofia SIP 'master' repo

[Antonis T. <ant...@te...>]

Thanks Moises for the quick answer. I found Sofia inside the Freeswitch
source, which is great, but I'm wondering about the following:

   - Can I somehow differentiate commits that have to do with Sofia? I see
   'mod_sofia' in some commits. Is that it? I guess I could probably come up
   with a git command to only show changes inside libs/sofia-sip, but
   wondering if the Freeswich folks used a convention
   - Can I find the last commit before Sofia was merged into Freeswitch
   project? So that I can follow history and see which changes where made to
   Sofia after the merge? Also which version of Sofia was Freeswitch basedon?
   Is it 1.12.11
   <https://gitorious.org/sofia-sip/sofia-sip?p=sofia-sip:sofia-sip.git;a=shortlog;h=refs/tags/1.12.11>
   ?
   - If I were to take Sofia from within the Freeswitch project and copy it
   as an independent bundle, would autotools work in
   configuring/making/installing etc? Or have dependencies been introduced
   over time and now Sofia needs to have Freeswitch in order to work?

Best regards and thanks in advance,
Antonis

On Mon, Jul 10, 2017 at 2:48 PM, Moises Silva <moi...@gm...>
wrote:

> On Mon, Jul 10, 2017 at 6:44 AM, Antonis Tsakiridis <
> ant...@te...> wrote:
>
>> Hello, is there a master repo for Sofia SIP that is considered active and
>> current? If so can you share where I can find it?
>>
>
> I think libsofia inside the FreeSWITCH git repo si the most up to date.
>



-- 
Antonis Tsakiridis
Lead Developer, Mobile SDKs at Telestax
ant...@te...
http://www.telestax.com

Re: [Sofia-sip-devel] Sofia SIP 'master' repo

[Moises S. <moi...@gm...>]

On Mon, Jul 10, 2017 at 6:44 AM, Antonis Tsakiridis <
ant...@te...> wrote:

> Hello, is there a master repo for Sofia SIP that is considered active and
> current? If so can you share where I can find it?
>

I think libsofia inside the FreeSWITCH git repo si the most up to date.

[Sofia-sip-devel] Sofia SIP 'master' repo

[Antonis T. <ant...@te...>]

Hello, is there a master repo for Sofia SIP that is considered active and
current? If so can you share where I can find it?

Thanks,
Antonis Tsakiridis

[Sofia-sip-devel] fail to make sofsip-cli-0.16 -- include files missing

[Ming C. <tom...@gm...>]

Hi,

I'm able to install sofia version 1.12.

But I got an error when build sofsip-cli-0.16:

 gcc -DHAVE_CONFIG_H -I. -I. -I.. -I/usr/local/include/sofia-sip-1.12
-I/usr/include/glib-2.0 -I/usr/lib/x86_64-linux-gnu/glib-2.0/include -g -O2
-MT ssc_sip.lo -MD -MP -MF .deps/ssc_sip.Tpo -c ssc_sip.c  -fPIC -DPIC -o
.libs/ssc_sip.o
ssc_sip.c:70:33: fatal error: sofia-sip/su_source.h: No such file or
directory
 #include <sofia-sip/su_source.h>

The failure is because include file su_source.h not found.

Why the 'make install' does not install include files correctly?

I do not see the su_source.h under
/usr/local/include/sofia-sip-1.12/sofia-sip directory, however, there are
other include files under that directory though.

Please help if you can.

thanks.
Ming




michen@michen-virtual-machine:/usr/local/include/sofia-sip-1.12$ ls
sofia-resolv  sofia-sip
michen@michen-virtual-machine:/usr/local/include/sofia-sip-1.12$

[Sofia-sip-devel] old sofsip-cli build errori

[Nenad M. <nmi...@ev...>]

Hi all,

Just started using sofia sip today and got stuck on compiling sofsip-cli-0.16 (15 has the same issue).  Sofia-sip I have installed is latest, 1.12.11.

Issue has been reported several times and it is about error:

/usr/local/include/sofia-sip-1.12/sofia-sip/soa_tag.h:120:3: error: 'soatag_local_sdp_str_ref' undeclared (first use in this function)
   soatag_local_sdp_str_ref, tag_str_vr(&(x))

, however, from old reports I could not figure out how to fix the problem.  Also, fix is supposed to be committed to some git repository that is unreachable for me.

If anybody still remembers this issue and knows how to fix it please share.

Thanks,

Nenad

Re: [Sofia-sip-devel] Reading custom SIP headers

[Andreas W. <a.w...@ya...>]

On 08/29/2016 08:32 AM, John Nash wrote:
> I am looking for some way to read value of a custom SIP header (starting
> with X-). I have access to object ssip_t. Which function I can use to
> extract custom SIP header and use its value?
>
>
>
> ------------------------------------------------------------------------------
>
>
> _______________________________________________
> Sofia-sip-devel mailing list
> Sof...@li...
> https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel

The sip structure contains a member of 'sip_unknown'.
You can use it like this to iterate custom (or "unknown") SIP headers:

sip_unknown_t* unknown_header = sip->sip_unknown;
while( unknown_header )
     {
         if( 0 == strcmp( unknown_header->un_name, "X-MyCustomHeader" ) )
         {
             /* whatever... */
         }

         unknown_header = unknown_header->un_next;
     }

Regards

[Sofia-sip-devel] Reading custom SIP headers

[John N. <joh...@gm...>]

I am looking for some way to read value of a custom SIP header (starting
with X-). I have access to object ssip_t. Which function I can use to
extract custom SIP header and use its value?

[Sofia-sip-devel] Current code

[<aik...@gm...>]

Good day, everyone!

I'm given a task to modify the sofia sip code (optimisations). So I'd like to work with the most current code. I've checked this list and seen a discussion in May about it, I understand that some day someone planned to make a central repository with all the changes that we brought to the code after the active development stopped.

But in fact, I'm not jealous to make these changes myself (they will be paid by the way), and I think it will be more feasible if someone already familiar with the code will do it. So if someone has a little time to do it, I'll happily hand you it:)

Therefore: is someone familiar with the code interested in doing this? Or if not can you help me in finding the latest code so I can work with it (and release the changes somewhere after)?

[Sofia-sip-devel] SIP REGISTER Contact header address dealing with NAT

[Antonis T. <ant...@te...>]

Hi all,

I'm working on a SIP client that uses Sofia SIP (behind NAT) and having
this issue where when registering to some SIP servers I am then unable to
receive incoming calls (for some others it works fine).

Seems that Sofia is populating the Contact header with a port different
than the port used as source port when sending the REGISTER. As a result
the incoming INVITE is destined to the Contact port and doesn't make it
through the NAT hole and the call is failed.

Any ideas how to remedy this? I have tried the following without success:
* Setting a specific port to bind to with NUTAG_URL, but I get the same
issue.
* Trying to get advantage of STUNTAG_SERVER() and STUNTAG_DOMAIN() Sofia
tags when creating NUA, but they don't seem to take any effect. Any ideas
how these are supposed to work roughly? Are they there to address issues
like the one I'm having and populating Contact header in the REGISTER with
an externally reachable ip:port?

Best regards.
Antonis Tsakiridis

Re: [Sofia-sip-devel] call-ID troubles

[Francesco L. <ali...@gm...>]

Hello Michael,
you mean the "tag" in the from header?
or the sofia tags?

here is the code used to send the re-invite

nua_invite(nh,

               TAG_IF(!proxyAddress.isEmpty(),
NUTAG_PROXY(proxyAddress.toLatin1().constData())),

               NUTAG_AUTOANSWER(0),

               TAG_IF(!options.isDataCall(),
SOATAG_USER_SDP_STR(sdp.toLatin1().constData())),

               SIPTAG_PRIORITY_STR(m_mapPriorityStrings[prio].toLatin1().constData()),

               TAG_END());


i am not manipulating the callid at all


On Wed, May 25, 2016 at 5:26 PM, Michael Jerris <mi...@je...> wrote:

> I would triple check the strings you are putting into the tags for this.
> I have never seen something like this unless it was actually me somehow
> buffer overflowing into it.
>
>
> On May 25, 2016, at 11:16 AM, Francesco Lamonica <ali...@gm...>
> wrote:
>
> Hello Michael,
> sorry, obviously i meant SIP :)
>
>
> On Sun, May 15, 2016 at 12:02 AM, Michael Jerris <mi...@je...> wrote:
>
>> Callid in sdp?
>>
>>
>> On Saturday, May 14, 2016, Francesco Lamonica <ali...@gm...>
>> wrote:
>>
>>> Hi all, i know this might be a shot in the dark however...
>>> i am experiencing a strange issue, randomically on a UAC that uses
>>> sofia-sip (latest stable from repository) when putting the remote on-hold.
>>> the scenario is the following
>>>
>>> sofia-uac -> invites
>>> 100
>>> 180
>>> 200
>>> sofia-uac -> on hold (changing the sdp manually and resending the invite)
>>> sofia-uac -> not on hold
>>>
>>> this scenario 1 out of 20/30 calls does a very strange thing
>>> the SDP that is sent out has the wrong callid so the remote party
>>> responds correctly with a series of 481
>>>
>>> the callid is mangled this way:
>>> the correct one is like 01234567890@172.16.17.15
>>> while the one sent is 01234567890@17217.117.15
>>> as if the second octet is overwritten by following parts of the address
>>>
>>> has anyone experienced something like that? or can give me some hints?
>>> thanks a lot
>>>
>>
>
>

Re: [Sofia-sip-devel] call-ID troubles

[Michael J. <mi...@je...>]

I would triple check the strings you are putting into the tags for this.  I have never seen something like this unless it was actually me somehow buffer overflowing into it.


> On May 25, 2016, at 11:16 AM, Francesco Lamonica <ali...@gm...> wrote:
> 
> Hello Michael,
> sorry, obviously i meant SIP :)
> 
> 
> On Sun, May 15, 2016 at 12:02 AM, Michael Jerris <mi...@je... <mailto:mi...@je...>> wrote:
> Callid in sdp?
> 
> 
> On Saturday, May 14, 2016, Francesco Lamonica <ali...@gm... <mailto:ali...@gm...>> wrote:
> Hi all, i know this might be a shot in the dark however...
> i am experiencing a strange issue, randomically on a UAC that uses sofia-sip (latest stable from repository) when putting the remote on-hold.
> the scenario is the following
> 
> sofia-uac -> invites
> 100
> 180
> 200
> sofia-uac -> on hold (changing the sdp manually and resending the invite)
> sofia-uac -> not on hold
> 
> this scenario 1 out of 20/30 calls does a very strange thing
> the SDP that is sent out has the wrong callid so the remote party responds correctly with a series of 481 
> 
> the callid is mangled this way:
> the correct one is like 01234567890@172.16.17.15 <>
> while the one sent is 01234567890@17217.117.15
> as if the second octet is overwritten by following parts of the address
> 
> has anyone experienced something like that? or can give me some hints?
> thanks a lot 
> 

Re: [Sofia-sip-devel] call-ID troubles

[Francesco L. <ali...@gm...>]

Hello Michael,
sorry, obviously i meant SIP :)


On Sun, May 15, 2016 at 12:02 AM, Michael Jerris <mi...@je...> wrote:

> Callid in sdp?
>
>
> On Saturday, May 14, 2016, Francesco Lamonica <ali...@gm...>
> wrote:
>
>> Hi all, i know this might be a shot in the dark however...
>> i am experiencing a strange issue, randomically on a UAC that uses
>> sofia-sip (latest stable from repository) when putting the remote on-hold.
>> the scenario is the following
>>
>> sofia-uac -> invites
>> 100
>> 180
>> 200
>> sofia-uac -> on hold (changing the sdp manually and resending the invite)
>> sofia-uac -> not on hold
>>
>> this scenario 1 out of 20/30 calls does a very strange thing
>> the SDP that is sent out has the wrong callid so the remote party
>> responds correctly with a series of 481
>>
>> the callid is mangled this way:
>> the correct one is like 01234567890@172.16.17.15
>> while the one sent is 01234567890@17217.117.15
>> as if the second octet is overwritten by following parts of the address
>>
>> has anyone experienced something like that? or can give me some hints?
>> thanks a lot
>>
>

Re: [Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

[Antonis T. <ant...@te...>]

Actually we have already ported Sofia for iOS and it's been working really
nice so far, combined with Webrtc for media ;). You can install it directly
on an iOS device from https://tsfr.io/xex2zc if you want to play with it,
and check code at https://github.com/RestComm/restcomm-ios-sdk -it's open
source. What we want to do though it to make integration easier for the iOS
Developer that wants to use just the Sofia part. Hence I said that we want
to come up with a more reusable API.

Anyway, thanks for for your input, we 'll be in touch :)

Antonis

On Fri, May 20, 2016 at 5:20 PM, Michael Jerris <mi...@je...> wrote:

> I haven't done any porting but seeing as this ha been well ported to
> symbian, I think from day one, it should be well suited for it.  All that
> should be down in su layer
>
>
> On Friday, May 20, 2016, Antonis Tsakiridis <
> ant...@te...> wrote:
>
>> Great, having an up to date standalone repo that interested contributors
>> can use will help a lot to keep Sofia alive and kicking :)
>>
>> Glad I found someone who knows his way around autotools, will ping you
>> when I start working for next iOS version to get this fixed; hopefully you
>> will have managed to create the repo by then so that I can contribute it
>> back ;)
>>
>> One more thing. Do you guys have experience porting Sofia on mobile
>> platforms? Would be a nice area to exchange experiences and help each
>> other. In Telestax we 're also thinking of coming up with a more reusable
>> API or even SDK over Sofia SIP, suitable for iOS developers that will make
>> it easy for community to take advantage of SIP functionality. Sofia has
>> proven great for our needs so far, but it's integration in iOS presents a
>> lot of challenges for newcomers. Would be nice to simplify that.
>>
>> Best regards,
>> Antonis
>>
>> On Thu, May 19, 2016 at 6:31 PM, Michael Jerris <mi...@je...> wrote:
>>
>>> The closest to a central maintained repo at this point is the copy in
>>> the freeswitch tree.  We will pull this out into a standalone repo sometime
>>> this summer before we do our next major version.  Unfortunately it seems
>>> like we are the last ones doing any maintaining of sofia-sip.  Making a way
>>> to choose between the two seems sane.  I'm probably the autotools guy who
>>> can help... Talk to me offline and I can look at the patches and send you
>>> in the right direction on how to resolve this.  The big issue with the
>>> sofia-sip in the freeswitch tree currently is that its set to build as a
>>> convenience lib not a normal libtool lib, I'll have to change that back and
>>> then make it an option too.
>>>
>>>
>>> On May 19, 2016, at 4:52 AM, Antonis Tsakiridis <
>>> ant...@te...> wrote:
>>>
>>> Hi Michael,
>>>
>>> First off, sorry for the late reply. Ok so what I did was to essentially
>>> replace openssl with boringssl in the auto tools build scripts and then
>>> allow Sofia to gather it's dependencies from the webrtc static lib that
>>> includes boringssl. Some points:
>>>
>>>    - Been trying to figure our a 'central' repository for Sofia SIP
>>>    code, so that I can contribute back to, but haven't been able to. I seem to
>>>    recall a gitorious repo but that doesn't seem to be around anymore. I see
>>>    that you have been involved with Sofia for the Freeswitch project, so maybe
>>>    you can give me some pointers?
>>>    - Before contributing back I'd like to improve my contribution so
>>>    that instead of replacing openssl with boringssl, to add it on top, so that
>>>    a developer can chose during configure time which of the two they need. The
>>>    problem is that I'm not that proficient with auto tools and it would take
>>>    me forever to do so. Do you have auto tools know-how? If so would you be
>>>    interested in contributing that part your self so that the community can
>>>    benefit from a more generic solution?
>>>
>>> Best regards,
>>> Antonis
>>>
>>> On Mon, Feb 15, 2016 at 5:31 PM, Michael Jerris <mi...@je...> wrote:
>>>
>>>> I'm interested
>>>>
>>>>
>>>> On Monday, February 15, 2016, Antonis Tsakiridis <
>>>> ant...@te...> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I just wanted to let you know that we 've been working with Sofia SIP
>>>>> and successfully combined it with Google's WebRTC native library for iOS
>>>>> and lately we managed to integrate the TLS part also.
>>>>>
>>>>> We did have some issues on the TLS integration due to the fact that
>>>>> Sofia is configured to work with OpenSSL, while WebRTC library already
>>>>> includes BoringSSL, and putting them all together was causing us trouble.
>>>>> So our solution was to change the auto tools configuration so that instead
>>>>> of OpenSSL it uses WebRTC library which includes Boring SSL and that did
>>>>> the trick. Would you be interested in such integration?
>>>>>
>>>>> Best regards
>>>>>
>>>>> --
>>>>> Antonis Tsakiridis
>>>>> Lead Developer, Mobile SDKs at TeleStax
>>>>> ant...@te...
>>>>> http://www.telestax.com
>>>>>
>>>>>
>>>
>>>
>>> --
>>> Antonis Tsakiridis
>>> Lead Developer, Mobile SDKs at TeleStax
>>> ant...@te...
>>> http://www.telestax.com
>>>
>>>
>>>
>>
>>
>> --
>> Antonis Tsakiridis
>> Lead Developer, Mobile SDKs at TeleStax
>> ant...@te...
>> http://www.telestax.com
>>
>>


-- 
Antonis Tsakiridis
Lead Developer, Mobile SDKs at TeleStax
ant...@te...
http://www.telestax.com

Re: [Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

[Michael J. <mi...@je...>]

I haven't done any porting but seeing as this ha been well ported to
symbian, I think from day one, it should be well suited for it.  All that
should be down in su layer

On Friday, May 20, 2016, Antonis Tsakiridis <ant...@te...>
wrote:

> Great, having an up to date standalone repo that interested contributors
> can use will help a lot to keep Sofia alive and kicking :)
>
> Glad I found someone who knows his way around autotools, will ping you
> when I start working for next iOS version to get this fixed; hopefully you
> will have managed to create the repo by then so that I can contribute it
> back ;)
>
> One more thing. Do you guys have experience porting Sofia on mobile
> platforms? Would be a nice area to exchange experiences and help each
> other. In Telestax we 're also thinking of coming up with a more reusable
> API or even SDK over Sofia SIP, suitable for iOS developers that will make
> it easy for community to take advantage of SIP functionality. Sofia has
> proven great for our needs so far, but it's integration in iOS presents a
> lot of challenges for newcomers. Would be nice to simplify that.
>
> Best regards,
> Antonis
>
> On Thu, May 19, 2016 at 6:31 PM, Michael Jerris <mi...@je...
> <javascript:_e(%7B%7D,'cvml','mi...@je...');>> wrote:
>
>> The closest to a central maintained repo at this point is the copy in the
>> freeswitch tree.  We will pull this out into a standalone repo sometime
>> this summer before we do our next major version.  Unfortunately it seems
>> like we are the last ones doing any maintaining of sofia-sip.  Making a way
>> to choose between the two seems sane.  I'm probably the autotools guy who
>> can help... Talk to me offline and I can look at the patches and send you
>> in the right direction on how to resolve this.  The big issue with the
>> sofia-sip in the freeswitch tree currently is that its set to build as a
>> convenience lib not a normal libtool lib, I'll have to change that back and
>> then make it an option too.
>>
>>
>> On May 19, 2016, at 4:52 AM, Antonis Tsakiridis <
>> ant...@te...
>> <javascript:_e(%7B%7D,'cvml','ant...@te...');>> wrote:
>>
>> Hi Michael,
>>
>> First off, sorry for the late reply. Ok so what I did was to essentially
>> replace openssl with boringssl in the auto tools build scripts and then
>> allow Sofia to gather it's dependencies from the webrtc static lib that
>> includes boringssl. Some points:
>>
>>    - Been trying to figure our a 'central' repository for Sofia SIP
>>    code, so that I can contribute back to, but haven't been able to. I seem to
>>    recall a gitorious repo but that doesn't seem to be around anymore. I see
>>    that you have been involved with Sofia for the Freeswitch project, so maybe
>>    you can give me some pointers?
>>    - Before contributing back I'd like to improve my contribution so
>>    that instead of replacing openssl with boringssl, to add it on top, so that
>>    a developer can chose during configure time which of the two they need. The
>>    problem is that I'm not that proficient with auto tools and it would take
>>    me forever to do so. Do you have auto tools know-how? If so would you be
>>    interested in contributing that part your self so that the community can
>>    benefit from a more generic solution?
>>
>> Best regards,
>> Antonis
>>
>> On Mon, Feb 15, 2016 at 5:31 PM, Michael Jerris <mi...@je...
>> <javascript:_e(%7B%7D,'cvml','mi...@je...');>> wrote:
>>
>>> I'm interested
>>>
>>>
>>> On Monday, February 15, 2016, Antonis Tsakiridis <
>>> ant...@te...
>>> <javascript:_e(%7B%7D,'cvml','ant...@te...');>>
>>> wrote:
>>>
>>>> Hello,
>>>>
>>>> I just wanted to let you know that we 've been working with Sofia SIP
>>>> and successfully combined it with Google's WebRTC native library for iOS
>>>> and lately we managed to integrate the TLS part also.
>>>>
>>>> We did have some issues on the TLS integration due to the fact that
>>>> Sofia is configured to work with OpenSSL, while WebRTC library already
>>>> includes BoringSSL, and putting them all together was causing us trouble.
>>>> So our solution was to change the auto tools configuration so that instead
>>>> of OpenSSL it uses WebRTC library which includes Boring SSL and that did
>>>> the trick. Would you be interested in such integration?
>>>>
>>>> Best regards
>>>>
>>>> --
>>>> Antonis Tsakiridis
>>>> Lead Developer, Mobile SDKs at TeleStax
>>>> ant...@te...
>>>> http://www.telestax.com
>>>>
>>>>
>>
>>
>> --
>> Antonis Tsakiridis
>> Lead Developer, Mobile SDKs at TeleStax
>> ant...@te...
>> <javascript:_e(%7B%7D,'cvml','ant...@te...');>
>> http://www.telestax.com
>>
>>
>>
>
>
> --
> Antonis Tsakiridis
> Lead Developer, Mobile SDKs at TeleStax
> ant...@te...
> <javascript:_e(%7B%7D,'cvml','ant...@te...');>
> http://www.telestax.com
>
>

Re: [Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

[Antonis T. <ant...@te...>]

Great, having an up to date standalone repo that interested contributors
can use will help a lot to keep Sofia alive and kicking :)

Glad I found someone who knows his way around autotools, will ping you when
I start working for next iOS version to get this fixed; hopefully you will
have managed to create the repo by then so that I can contribute it back ;)

One more thing. Do you guys have experience porting Sofia on mobile
platforms? Would be a nice area to exchange experiences and help each
other. In Telestax we 're also thinking of coming up with a more reusable
API or even SDK over Sofia SIP, suitable for iOS developers that will make
it easy for community to take advantage of SIP functionality. Sofia has
proven great for our needs so far, but it's integration in iOS presents a
lot of challenges for newcomers. Would be nice to simplify that.

Best regards,
Antonis

On Thu, May 19, 2016 at 6:31 PM, Michael Jerris <mi...@je...> wrote:

> The closest to a central maintained repo at this point is the copy in the
> freeswitch tree.  We will pull this out into a standalone repo sometime
> this summer before we do our next major version.  Unfortunately it seems
> like we are the last ones doing any maintaining of sofia-sip.  Making a way
> to choose between the two seems sane.  I'm probably the autotools guy who
> can help... Talk to me offline and I can look at the patches and send you
> in the right direction on how to resolve this.  The big issue with the
> sofia-sip in the freeswitch tree currently is that its set to build as a
> convenience lib not a normal libtool lib, I'll have to change that back and
> then make it an option too.
>
>
> On May 19, 2016, at 4:52 AM, Antonis Tsakiridis <
> ant...@te...> wrote:
>
> Hi Michael,
>
> First off, sorry for the late reply. Ok so what I did was to essentially
> replace openssl with boringssl in the auto tools build scripts and then
> allow Sofia to gather it's dependencies from the webrtc static lib that
> includes boringssl. Some points:
>
>    - Been trying to figure our a 'central' repository for Sofia SIP code,
>    so that I can contribute back to, but haven't been able to. I seem to
>    recall a gitorious repo but that doesn't seem to be around anymore. I see
>    that you have been involved with Sofia for the Freeswitch project, so maybe
>    you can give me some pointers?
>    - Before contributing back I'd like to improve my contribution so that
>    instead of replacing openssl with boringssl, to add it on top, so that a
>    developer can chose during configure time which of the two they need. The
>    problem is that I'm not that proficient with auto tools and it would take
>    me forever to do so. Do you have auto tools know-how? If so would you be
>    interested in contributing that part your self so that the community can
>    benefit from a more generic solution?
>
> Best regards,
> Antonis
>
> On Mon, Feb 15, 2016 at 5:31 PM, Michael Jerris <mi...@je...> wrote:
>
>> I'm interested
>>
>>
>> On Monday, February 15, 2016, Antonis Tsakiridis <
>> ant...@te...> wrote:
>>
>>> Hello,
>>>
>>> I just wanted to let you know that we 've been working with Sofia SIP
>>> and successfully combined it with Google's WebRTC native library for iOS
>>> and lately we managed to integrate the TLS part also.
>>>
>>> We did have some issues on the TLS integration due to the fact that
>>> Sofia is configured to work with OpenSSL, while WebRTC library already
>>> includes BoringSSL, and putting them all together was causing us trouble.
>>> So our solution was to change the auto tools configuration so that instead
>>> of OpenSSL it uses WebRTC library which includes Boring SSL and that did
>>> the trick. Would you be interested in such integration?
>>>
>>> Best regards
>>>
>>> --
>>> Antonis Tsakiridis
>>> Lead Developer, Mobile SDKs at TeleStax
>>> ant...@te...
>>> http://www.telestax.com
>>>
>>>
>
>
> --
> Antonis Tsakiridis
> Lead Developer, Mobile SDKs at TeleStax
> ant...@te...
> http://www.telestax.com
>
>
>


-- 
Antonis Tsakiridis
Lead Developer, Mobile SDKs at TeleStax
ant...@te...
http://www.telestax.com

Re: [Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

[Michael J. <mi...@je...>]

The closest to a central maintained repo at this point is the copy in the freeswitch tree.  We will pull this out into a standalone repo sometime this summer before we do our next major version.  Unfortunately it seems like we are the last ones doing any maintaining of sofia-sip.  Making a way to choose between the two seems sane.  I'm probably the autotools guy who can help... Talk to me offline and I can look at the patches and send you in the right direction on how to resolve this.  The big issue with the sofia-sip in the freeswitch tree currently is that its set to build as a convenience lib not a normal libtool lib, I'll have to change that back and then make it an option too.


> On May 19, 2016, at 4:52 AM, Antonis Tsakiridis <ant...@te...> wrote:
> 
> Hi Michael, 
> 
> First off, sorry for the late reply. Ok so what I did was to essentially replace openssl with boringssl in the auto tools build scripts and then allow Sofia to gather it's dependencies from the webrtc static lib that includes boringssl. Some points:
> Been trying to figure our a 'central' repository for Sofia SIP code, so that I can contribute back to, but haven't been able to. I seem to recall a gitorious repo but that doesn't seem to be around anymore. I see that you have been involved with Sofia for the Freeswitch project, so maybe you can give me some pointers?
> Before contributing back I'd like to improve my contribution so that instead of replacing openssl with boringssl, to add it on top, so that a developer can chose during configure time which of the two they need. The problem is that I'm not that proficient with auto tools and it would take me forever to do so. Do you have auto tools know-how? If so would you be interested in contributing that part your self so that the community can benefit from a more generic solution?
> Best regards,
> Antonis
> 
> On Mon, Feb 15, 2016 at 5:31 PM, Michael Jerris <mi...@je... <mailto:mi...@je...>> wrote:
> I'm interested
> 
> 
> On Monday, February 15, 2016, Antonis Tsakiridis <ant...@te... <mailto:ant...@te...>> wrote:
> Hello, 
> 
> I just wanted to let you know that we 've been working with Sofia SIP and successfully combined it with Google's WebRTC native library for iOS and lately we managed to integrate the TLS part also. 
> 
> We did have some issues on the TLS integration due to the fact that Sofia is configured to work with OpenSSL, while WebRTC library already includes BoringSSL, and putting them all together was causing us trouble. So our solution was to change the auto tools configuration so that instead of OpenSSL it uses WebRTC library which includes Boring SSL and that did the trick. Would you be interested in such integration?
> 
> Best regards 
> 
> -- 
> Antonis Tsakiridis
> Lead Developer, Mobile SDKs at TeleStax
> ant...@te... <>
> http://www.telestax.com <http://www.telestax.com/>
> 
> 
> 
> 
> -- 
> Antonis Tsakiridis
> Lead Developer, Mobile SDKs at TeleStax
> ant...@te... <mailto:ant...@te...>
> http://www.telestax.com <http://www.telestax.com/>
> 

Re: [Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

[Antonis T. <ant...@te...>]

Hi Michael,

First off, sorry for the late reply. Ok so what I did was to essentially
replace openssl with boringssl in the auto tools build scripts and then
allow Sofia to gather it's dependencies from the webrtc static lib that
includes boringssl. Some points:

   - Been trying to figure our a 'central' repository for Sofia SIP code,
   so that I can contribute back to, but haven't been able to. I seem to
   recall a gitorious repo but that doesn't seem to be around anymore. I see
   that you have been involved with Sofia for the Freeswitch project, so maybe
   you can give me some pointers?
   - Before contributing back I'd like to improve my contribution so that
   instead of replacing openssl with boringssl, to add it on top, so that a
   developer can chose during configure time which of the two they need. The
   problem is that I'm not that proficient with auto tools and it would take
   me forever to do so. Do you have auto tools know-how? If so would you be
   interested in contributing that part your self so that the community can
   benefit from a more generic solution?

Best regards,
Antonis

On Mon, Feb 15, 2016 at 5:31 PM, Michael Jerris <mi...@je...> wrote:

> I'm interested
>
>
> On Monday, February 15, 2016, Antonis Tsakiridis <
> ant...@te...> wrote:
>
>> Hello,
>>
>> I just wanted to let you know that we 've been working with Sofia SIP and
>> successfully combined it with Google's WebRTC native library for iOS and
>> lately we managed to integrate the TLS part also.
>>
>> We did have some issues on the TLS integration due to the fact that Sofia
>> is configured to work with OpenSSL, while WebRTC library already includes
>> BoringSSL, and putting them all together was causing us trouble. So our
>> solution was to change the auto tools configuration so that instead of
>> OpenSSL it uses WebRTC library which includes Boring SSL and that did the
>> trick. Would you be interested in such integration?
>>
>> Best regards
>>
>> --
>> Antonis Tsakiridis
>> Lead Developer, Mobile SDKs at TeleStax
>> ant...@te...
>> http://www.telestax.com
>>
>>


-- 
Antonis Tsakiridis
Lead Developer, Mobile SDKs at TeleStax
ant...@te...
http://www.telestax.com

Re: [Sofia-sip-devel] call-ID troubles

[Michael J. <mi...@je...>]

Callid in sdp?

On Saturday, May 14, 2016, Francesco Lamonica <ali...@gm...>
wrote:

> Hi all, i know this might be a shot in the dark however...
> i am experiencing a strange issue, randomically on a UAC that uses
> sofia-sip (latest stable from repository) when putting the remote on-hold.
> the scenario is the following
>
> sofia-uac -> invites
> 100
> 180
> 200
> sofia-uac -> on hold (changing the sdp manually and resending the invite)
> sofia-uac -> not on hold
>
> this scenario 1 out of 20/30 calls does a very strange thing
> the SDP that is sent out has the wrong callid so the remote party responds
> correctly with a series of 481
>
> the callid is mangled this way:
> the correct one is like 01234567890@172.16.17.15
> <javascript:_e(%7B%7D,'cvml','01234567890@172.16.17.15');>
> while the one sent is 01234567890@17217.117.15
> as if the second octet is overwritten by following parts of the address
>
> has anyone experienced something like that? or can give me some hints?
> thanks a lot
>

[Sofia-sip-devel] call-ID troubles

[Francesco L. <ali...@gm...>]

Hi all, i know this might be a shot in the dark however...
i am experiencing a strange issue, randomically on a UAC that uses
sofia-sip (latest stable from repository) when putting the remote on-hold.
the scenario is the following

sofia-uac -> invites
100
180
200
sofia-uac -> on hold (changing the sdp manually and resending the invite)
sofia-uac -> not on hold

this scenario 1 out of 20/30 calls does a very strange thing
the SDP that is sent out has the wrong callid so the remote party responds
correctly with a series of 481

the callid is mangled this way:
the correct one is like 01234567890@172.16.17.15
while the one sent is 01234567890@17217.117.15
as if the second octet is overwritten by following parts of the address

has anyone experienced something like that? or can give me some hints?
thanks a lot

Re: [Sofia-sip-devel] How can I ignore an incoming call using Sofia SIP

[Antonis T. <ant...@te...>]

Thanks a lot Saiyam for your reply. Sorry for taking so long to write back,
but this somehow ended up in Spam :(

Anyway, the problem with your approach is that I don't want Sofia to
respond at all. That is I don't want a 403 Forbidden to be issued.

The use case I'm trying to cover is similar to what mobile phones provide
when you want to ignore a call. In that scenario the caller thinks the call
is still ringing, but to the receiver it is ignored. The problem is that
when I tried to enforce that by just deleting the Sofia handle, I see that
Sofia internally still prepares and sends a response, which is what I'm
trying to avoid.

Any other clues?

Best regards,
Antonis

On Tue, Feb 2, 2016 at 7:49 PM, Saiyam Doshi <sai...@ya...> wrote:

> Hi Antonis,
>
> You can ignore incoming invite by setting nua parameters. Just provide SIP
> method names which you would like to allow for your SIP client application
> by passing them in "SIPTAG_ALLOW_STR".
>
> For example,
>
> After call to nua_create call below API.
>
> nua_set_params(ptrApp->AppCtx.Nua,
>                 SIPTAG_ALLOW_STR("BYE,CANCEL,OPTIONS,INFO,MESSAGE"), //
> don't provide INVITE here
>                 NUTAG_ENABLEMESSAGE(1),
>                 NUTAG_ENABLEINVITE(0), // set 0 here
>                 TAG_NULL());
>
> By setting this Incoming INVITE is disabled and NUA answers 403 Forbidden.
>
> - saiyam
>
>
> On Tuesday, February 2, 2016 7:18 PM, Antonis Tsakiridis <
> ant...@te...> wrote:
>
>
> By ignore I mean to tell the local SIP stack to remove all resources for
> the incoming call, but not send any SIP message across.
>
> I've trying do that using nua_handle_destroy() for the unanswered sofia
> handle (i.e. without sending BYE, or answering with an 'error' code), but
> it seems that Sofia generates a response anyway.
>
> Is there a way to avoid this? I've been able to do that using JAIN SIP
> stack but not Sofia.
>
> Best regards,
> Antonis
>
>
> ------------------------------------------------------------------------------
> Site24x7 APM Insight: Get Deep Visibility into Application Performance
> APM + Mobile APM + RUM: Monitor 3 App instances at just $35/Month
> Monitor end-to-end web transactions and take corrective actions now
> Troubleshoot faster and improve end-user experience. Signup Now!
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> _______________________________________________
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>
>
>


-- 
Antonis Tsakiridis
Lead Developer, Mobile SDKs at TeleStax
ant...@te...
http://www.telestax.com

[Sofia-sip-devel] [FIX] crash when querying adapter addresses failed (Windows)

[Andreas W. <a.w...@ya...>]

During long and heavy testruns I encountered seemingly random crashes 
running on Windows 7.
I tracked the problem down to the function

static int win_localinfo(su_localinfo_t const hints[1], su_localinfo_t 
**rresult);

which resides in su_localinfo.c.

If the call to GetAdapterAddresses() fails, the function tries to use 
FormatMessage()
to get an error string to print out.
Unfortunately, the call to FormatMessage() specifies an illegal buffer:

if (!FormatMessage(FORMAT_MESSAGE_ALLOCATE_BUFFER |
                        FORMAT_MESSAGE_FROM_SYSTEM |
                        FORMAT_MESSAGE_IGNORE_INSERTS,
                        NULL,
                        error,
                        MAKELANGID(LANG_NEUTRAL, SUBLANG_DEFAULT),
                        msg, 0, NULL))
     msg = empty;

It SHOULD be:

if (!FormatMessage(FORMAT_MESSAGE_ALLOCATE_BUFFER |
                        FORMAT_MESSAGE_FROM_SYSTEM |
                        FORMAT_MESSAGE_IGNORE_INSERTS,
                        NULL,
                        error,
                        MAKELANGID(LANG_NEUTRAL, SUBLANG_DEFAULT),
                        (LPTSTR)&msg, 0, NULL))
     msg = empty;

Earlier in the function it says:

char const* empty = "";
LPTSTR msg = empty;

Calling FormatMessage() and specifying a pointer to empty (assignment)
is obviously illegal as it requires the address to a pointer.


Best regards,
Andreas

Re: [Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

[Michael J. <mi...@je...>]

I'm interested

On Monday, February 15, 2016, Antonis Tsakiridis <
ant...@te...> wrote:

> Hello,
>
> I just wanted to let you know that we 've been working with Sofia SIP and
> successfully combined it with Google's WebRTC native library for iOS and
> lately we managed to integrate the TLS part also.
>
> We did have some issues on the TLS integration due to the fact that Sofia
> is configured to work with OpenSSL, while WebRTC library already includes
> BoringSSL, and putting them all together was causing us trouble. So our
> solution was to change the auto tools configuration so that instead of
> OpenSSL it uses WebRTC library which includes Boring SSL and that did the
> trick. Would you be interested in such integration?
>
> Best regards
>
> --
> Antonis Tsakiridis
> Lead Developer, Mobile SDKs at TeleStax
> ant...@te...
> <javascript:_e(%7B%7D,'cvml','ant...@te...');>
> http://www.telestax.com
>
>

[Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

[Antonis T. <ant...@te...>]

Hello,

I just wanted to let you know that we 've been working with Sofia SIP and
successfully combined it with Google's WebRTC native library for iOS and
lately we managed to integrate the TLS part also.

We did have some issues on the TLS integration due to the fact that Sofia
is configured to work with OpenSSL, while WebRTC library already includes
BoringSSL, and putting them all together was causing us trouble. So our
solution was to change the auto tools configuration so that instead of
OpenSSL it uses WebRTC library which includes Boring SSL and that did the
trick. Would you be interested in such integration?

Best regards

-- 
Antonis Tsakiridis
Lead Developer, Mobile SDKs at TeleStax
ant...@te...
http://www.telestax.com

Re: [Sofia-sip-devel] How can I ignore an incoming call using Sofia SIP

[Saiyam D. <sai...@ya...>]

Hi Antonis,
You can ignore incoming invite by setting nua parameters. Just provide SIP method names which you would like to allow for your SIP client application by passing them in "SIPTAG_ALLOW_STR".
For example,
After call to nua_create call below API.
nua_set_params(ptrApp->AppCtx.Nua,                SIPTAG_ALLOW_STR("BYE,CANCEL,OPTIONS,INFO,MESSAGE"), // don't provide INVITE here                NUTAG_ENABLEMESSAGE(1),                NUTAG_ENABLEINVITE(0), // set 0 here                TAG_NULL());
By setting this Incoming INVITE is disabled and NUA answers 403 Forbidden. - saiyam 

    On Tuesday, February 2, 2016 7:18 PM, Antonis Tsakiridis <ant...@te...> wrote:
 

 By ignore I mean to tell the local SIP stack to remove all resources for the incoming call, but not send any SIP message across.

I've trying do that using nua_handle_destroy() for the unanswered sofia handle (i.e. without sending BYE, or answering with an 'error' code), but it seems that Sofia generates a response anyway.

Is there a way to avoid this? I've been able to do that using JAIN SIP stack but not Sofia.

Best regards, Antonis
------------------------------------------------------------------------------
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APM + Mobile APM + RUM: Monitor 3 App instances at just $35/Month
Monitor end-to-end web transactions and take corrective actions now
Troubleshoot faster and improve end-user experience. Signup Now!
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_______________________________________________
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Sof...@li...
https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel


  

[Sofia-sip-devel] How can I ignore an incoming call using Sofia SIP

[Antonis T. <ant...@te...>]

By ignore I mean to tell the local SIP stack to remove all resources for
the incoming call, but not send any SIP message across.

I've trying do that using nua_handle_destroy() for the unanswered sofia
handle (i.e. without sending BYE, or answering with an 'error' code), but
it seems that Sofia generates a response anyway.

Is there a way to avoid this? I've been able to do that using JAIN SIP
stack but not Sofia.

Best regards,
Antonis